[Asterisk-Users] Sip Trunking

[Steven Critchfield]

On Mon, 2004-01-05 at 12:57, Eduardo Goncalves wrote:
> On Mon, 5 Jan 2004 11:20:08 -0600 (CST)
> Brian West <brian at bkw.org> wrote:
> 
> > Why not use IAX2 trunking you can accomplish the same results with ..
> > I tried SIP to SIP with asterisk you must do it without passwords.
> 
> 	Because cisco doesn't compress IAX headers, only rtp.

Have you verified that IAX2 trunking doesn't save you the same or more
bandwidth, or are you stuck on your solution of using the Ciscos? I
think IAX2 is already a small header, and trunking saves you 8 bytes per
packet per call over 1. So at 4 calls you have 32 bytes * 50 per second
of saving. Thats 1800 bytes per second. 

> > On Mon, 5 Jan 2004, Eduardo Goncalves wrote:
> > 
> > > Hi list,
> > >
> > > 	I have to connect two asterisk box, in this scenario:
> > >
> > > [asterisk1]----sip----[asterisk2]----PSTN
> > >
> > > 	I must use sip, cos we'll use cisco rtp header-compression to
> > > 	save
> > > bandwidth.
> > >
> > > 	Could you tell me the best way to send calls from asterisk1 to
> > > asterisk2, since I cannot use IAX trunking?
> > >
> > > Thanks in advance
> > > Eduardo

-- 
Steven Critchfield <critch at basesys.com>